I have an Asterisk serving 15 people using the X-Lite soft-phone. Currently they all register to the internal IP address of Asterisk (192.168.1.110). I only use VoIP internally. External calls go PSTN. I'd like to arrange it so that they register to our external WAN address (port forwarded to Asterisk) so that they can go mobile and still have Asterisk service. Is it possible to arrange it so that when in the office, the SIP signaling goes through the external WAN, but the Media (Voice) traffic stays local? In other words when a user is on the local LAN, I don't want their voice traffic going out on the net and then back in. Thanks, Hugh
hugolivude
2005-Aug-11 12:28 UTC
[Asterisk-Users] Re: SIP signaling vs Media (Voice) Traffic
Thanks to JT for the response below: The short answer is: no. However, there is a solution. Set up a nameserver in your office that replies with the "internal" address of your Asterisk server to systems that are on your office LAN, and replies with the "outside" address of your Asterisk server when asked from hosts outside your LAN. This can be done crudely with just launching a local version of BIND that has a different zone file, or it can be done more elegantly with DNS "views" on your primary nameserver. I'll let you and Google figure out how to do it. :-) On 8/5/05, hugolivude <hugolivude@gmail.com> wrote:> I have an Asterisk serving 15 people using the X-Lite soft-phone. > Currently they all register to the internal IP address of Asterisk > (192.168.1.110). I only use VoIP internally. External calls go PSTN. > > I'd like to arrange it so that they register to our external WAN > address (port forwarded to Asterisk) so that they can go mobile and > still have Asterisk service. > > Is it possible to arrange it so that when in the office, the SIP > signaling goes through the external WAN, but the Media (Voice) traffic > stays local? In other words when a user is on the local LAN, I don't > want their voice traffic going out on the net and then back in. > > Thanks, > Hugh >
gw@adcomcorp.com
2005-Aug-11 19:17 UTC
[Asterisk-Users] Re: SIP signaling vs Media (Voice) Traffic
15 people? Why not use a hosts file? Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of hugolivude Sent: Thursday, August 11, 2005 3:29 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Re: SIP signaling vs Media (Voice) Traffic Thanks to JT for the response below: The short answer is: no. However, there is a solution. Set up a nameserver in your office that replies with the "internal" address of your Asterisk server to systems that are on your office LAN, and replies with the "outside" address of your Asterisk server when asked from hosts outside your LAN. This can be done crudely with just launching a local version of BIND that has a different zone file, or it can be done more elegantly with DNS "views" on your primary nameserver. I'll let you and Google figure out how to do it. :-) On 8/5/05, hugolivude <hugolivude@gmail.com> wrote:> I have an Asterisk serving 15 people using the X-Lite soft-phone. > Currently they all register to the internal IP address of Asterisk > (192.168.1.110). I only use VoIP internally. External calls go PSTN. > > I'd like to arrange it so that they register to our external WAN > address (port forwarded to Asterisk) so that they can go mobile and > still have Asterisk service. > > Is it possible to arrange it so that when in the office, the SIP > signaling goes through the external WAN, but the Media (Voice) traffic> stays local? In other words when a user is on the local LAN, I don't > want their voice traffic going out on the net and then back in. > > Thanks, > Hugh >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users